1. Field of the Invention
The present invention relates to a speech processing apparatus in a speech coding apparatus, speech decoding apparatus, speech reproducing apparatus, or the like for improving the intelligibility of a speech signal degraded in quality or enhancing input speech so as to enable output speech to be intelligibly heard even in a noisy environment or other environment where the speech is difficult to understand and a mobile phone or other mobile communication terminal provided with such a speech processing apparatus.
2. Description of the Related Art
Various technologies exists for processing speech signals to improve the intelligibility of speech degraded in quality and difficult to understand. For example, numerous systems have been proposed and applied to mobile phones for so-called “noise cancelers” for removing noise mixed in with speech.
Mobile phones etc. are often used in noisy environments. When using mobile phones in noisy environments, there is the problem that the other party is difficult to understand. Therefore, various technologies have been proposed to enable speech to be easily understood by processing for enhancing the characteristics of the speech.
For example, as a technique for enhancing the formants, important for vowel recognition of speech, Japanese Unexamined Patent Publication (Kokai) No. 2-82710 has proposed technology using a post-processing filter having a transfer characteristic H(z) expressed by the following equation (1):H(z)={Σi=1na[i](βz)−1}/{Σi=1ma[i](αz)−1}  (1)
In the above equation (1), “a[i]” is a linear prediction coefficient (LPC), while α and β are suitably determined constants. By using a post-processing filter having a characteristic expressed by the above equation (1), the formant frequency component is enhanced and the subjective quality of the encoded speech is improved.
Further, various technologies have been proposed for formant enhancement using line spectrum pairs (LSPs). An LSP is a frequency parameter expressing the characteristics of speech. If expressing an LSP by the variable ω, ω is usually in the range of 0≦ω≦π, but depending on the method of expression, it is sometimes also expressed by a range normalized to a value between 0 and 1, that is, 0≦ω≦1. Alternatively, it is sometimes expressed as 0≦ω≦4000 (Hz). Further, the cosine of an LSP, that is, cos(ω), is also called an “LSP”. An LSP can be calculated by computation from an LPC. Further, an LPC can be calculated from an LSP.
By setting as the LSPs values increasing steadily from a low order to a high order, it is known that the later filtering proceeds stably. Further, the smaller the distance (difference) between LSP values of adjacent orders, the stronger the peak that appears in the formants of the speech. This property becomes greater the closer the value of an LSP to 0. LSPs are for example explained in detail in for example the Acoustic Society of Japan, “Oto no Komunikeesyon Kogaku” (Communication Engineering of Sound), first edition, Corona, Aug. 30, 1996, p. 27.
Japanese Unexamined Patent Publication (Kokai) No. 8-305397 proposes a speech processing filter calculating an interior division value with predetermined LSP values (values arranged at equal intervals on the frequency) for input values of LSPs, making corrections to widen portions where the distance between adjacent orders is less than a predetermined value, and increasing the freedom of characteristics of the speech processing filter and obtaining an excellent formant enhancement effect without causing distortion of the level of perception in the range of the permissible spectral gradients.
Japanese Unexamined Patent Publication (Kokai) No. 2000-242298 proposes an LSP correction device which uses an ascending order LSP corrector which calculates the distance between adjacent orders successively from the lower order of the LSPs and widens the distance between orders when the distance between orders falls below a threshold and a descending order LSP corrector which calculates the distance between adjacent orders successively from the higher order of the LSPs and widens the distance between orders when the distance between orders falls below a threshold so as to enable the distance between orders to be sufficiently widened with a good balance.
The above related art, however, suffered from the following problems.
In the post-processing filter of Japanese Unexamined Patent Publication (Kokai) No. 2-82710, it was necessary to adjust the constant parameters α and β. These parameters, however, are difficult to adjust since it is difficult to determine the correspondence between frequency characteristics and auditory effects. If unsuitably adjusted, the sound quality conversely ends up deteriorating.
Further, in the speech processing filter of Japanese Unexamined Patent Publication (Kokai) No. 8-305397, since the correction is made by obtaining the interior division point between the LSP values of the speech signal and LSP values arranged at equal intervals in advance, when the original LSP values concentrate at a lower band, the speech ends up shifting to a high frequency overall and the output speech is liable to sound strange.
Further, in the LSP correction device of Japanese Unexamined Patent Publication (Kokai) No. 2000-242298, since the LSP values of adjacent orders are successively changed, when there is unevenness in the original distribution of the LSPs, trouble such as the LSP values ending up leaning heavily to the low order or high order side is liable to occur.